[asterisk-bugs] [Asterisk 0018601]: No RTP port update when SIP RE-INVITE is received

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=18601 
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Reported By:                kondik
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18601
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.2.15 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             2011-01-11 16:10 CST
Last Modified:              2011-01-19 14:39 CST
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Summary:                    No RTP port update when SIP RE-INVITE is received
Description: 
Exactly the same issue as: https://issues.asterisk.org/view.php?id=15149
but those issue is closed, so I open new issue.

When RE-INVITE comes to asterisk with new media port, asterisk sends 200
OK, and send RTP packets further to the old port instead of new port which
received with RE-INVITE.

My asterisk version is: 1.6.2.15

Could you help me with this?
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Relationships       ID      Summary
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has duplicate       0015149 No audio on SIP RE-INVITE connecting wi...
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Issue History 
Date Modified    Username       Field                    Change               
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2011-01-19 14:39 lmadsen        Resolution               open => no change
required
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